Hi chris. The result it is the same, no sound. -- Executing Answer("SIP/7010-081f6f68", "") in new stack -- Executing Playback("SIP/7010-081f6f68", "beep") in new stack -- Playing 'beep' (language 'en')
more sugestion? 2007/4/18, Christopher Aloi <[EMAIL PROTECTED]>:
Try getting rid of all those macros etc.. so you can see what's going on, something simple like: exten => 500,1,Answer() exten => 500,n,Playback(beep) exten => 500,n,Hangup() Then dial 500 from your soft phone and see what happens. On 4/17/07, EWV2 <[EMAIL PROTECTED]> wrote: > The codecs are correct, so you are having other type of problem > > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Carlos > Jerónimo > Sent: Tuesday, April 17, 2007 5:10 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] internal sounds of asterisk / freePBX > > HI, my sip.conf /codecs > > disallow=all > allow=ulaw > allow=alaw > > this codcs is correct? > thanks > > > > 2007/4/17, EWV2 <[EMAIL PROTECTED]>: > > It sounds like a codec problem. > > > > What codec are you using? > > > > If you are using g723.1 or g729 passthru you will not be able to hear > > nothing > > > > > > -----Original Message----- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of Carlos > > Jerónimo > > Sent: Tuesday, April 17, 2007 4:30 PM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: [asterisk-users] internal sounds of asterisk / freePBX > > > > Sorry but i can't register in the freepbx forum, so this is my > > solutons for resolve my trouble. > > > > HI, my problem is with internal sounds of asterisk. > > for example when calling voicemail, no system recordings are being > > played back. However, when running asterisk > > in a debug mode, i see the call coming through to the system and the > > system playing back the wav files promptly. > > However, no sound comes through. I have verified that the sounds are > > in the correct location and that > > asterisk:asterisk has access to all files, is music on hold works, but > > other than that no system recordings are audible. > > > > But this isn't just voicemail. It's every system recording. Such as > > the feature code *60 to > > play the current time. It shows the call connected and it shows to be > > playing the wav file, but nothing > > coming out of the speaker of the phone....didn't just try with one phone > > either > > > > In other words, asterisk shows it's all working well. my logs: > > > > == Spawn extension (macro-systemrecording, h, 1) exited non-zero on > > 'SIP/7010-081d7288' > > -- Executing Macro("SIP/7010-0819b350", "user-callerid|") in new stack > > -- Executing NoOp("SIP/7010-0819b350", "user-callerid: device > > 7010") in new stack > > -- Executing GotoIf("SIP/7010-0819b350", "0?report") in new stack > > -- Executing GotoIf("SIP/7010-0819b350", "0?start") in new stack > > -- Executing Set("SIP/7010-0819b350", "REALCALLERIDNUM=7010") in new > > stack > > -- Executing NoOp("SIP/7010-0819b350", "REALCALLERIDNUM is 7010") > > in new stack > > -- Executing Set("SIP/7010-0819b350", "AMPUSER=7010") in new stack > > -- Executing Set("SIP/7010-0819b350", "AMPUSERCIDNAME=Portaria") > > in new stack > > -- Executing GotoIf("SIP/7010-0819b350", "0?report") in new stack > > -- Executing Set("SIP/7010-0819b350", "CALLERID(all)=Portaria > > <7010>") in new stack > > -- Executing Set("SIP/7010-0819b350", "REALCALLERIDNUM=7010") in new > > stack > > -- Executing NoOp("SIP/7010-0819b350", "TTL: ARG1: ") in new stack > > -- Executing GotoIf("SIP/7010-0819b350", "0?continue") in new stack > > -- Executing Set("SIP/7010-0819b350", "_TTL=64") in new stack > > -- Executing GotoIf("SIP/7010-0819b350", "1?continue") in new stack > > -- Goto (macro-user-callerid,s,21) > > -- Executing NoOp("SIP/7010-0819b350", "Using CallerID "Portaria" > > <7010>") in new stack > > -- Executing Wait("SIP/7010-0819b350", "2") in new stack > > -- Executing Macro("SIP/7010-0819b350", > > "systemrecording|dorecord") in new stack > > -- Executing Goto("SIP/7010-0819b350", "dorecord|1") in new stack > > -- Goto (macro-systemrecording,dorecord,1) > > -- Executing Record("SIP/7010-0819b350", > > "/tmp/7010-ivrrecording:wav") in new stack > > -- Playing 'beep' (language 'en') > > > > Really at a stand still until I can get this resolved so any thoughts > > are much appreciated. > > > > > > -- > > Carlos Jerónimo > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > -- > Carlos Jerónimo > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- ------ Christopher T Aloi ------ _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
-- Carlos Jerónimo _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users